> ## Documentation Index
> Fetch the complete documentation index at: https://docs.aisky.co.za/llms.txt
> Use this file to discover all available pages before exploring further.

# Asterisk

> Connect raw Asterisk (pjsip) to AiSky. For teams running Asterisk without a GUI.

<Info>
  **Audience**: Asterisk admin comfortable editing `pjsip.conf`. **Time**: 10 minutes. **Driver**: chan\_pjsip.
</Info>

## Prerequisites

* Asterisk 16 or later (17, 18, 20, 21 all work)
* chan\_pjsip loaded (`pjsip show version`)
* Outbound UDP 5060 allowed
* SIP ALG disabled upstream

## Step 1: create trunk in AiSky

SIP Trunk wizard, pick **Asterisk**. Save and copy credentials.

## Step 2: pjsip.conf configuration

Append to `/etc/asterisk/pjsip.conf` (or use a separate conf file):

```
[smartalex-transport]
type=transport
protocol=udp
bind=0.0.0.0

[smartalex-registration]
type=registration
outbound_auth=smartalex-auth
server_uri=sip:sip.voice.getsmartalex.com
client_uri=sip:YOUR_USERNAME@sip.voice.getsmartalex.com
retry_interval=60
expiration=3600

[smartalex-auth]
type=auth
auth_type=userpass
username=YOUR_USERNAME
password=YOUR_PASSWORD

[smartalex]
type=aor
contact=sip:sip.voice.getsmartalex.com

[smartalex]
type=endpoint
transport=smartalex-transport
outbound_auth=smartalex-auth
aors=smartalex
context=from-smartalex
disallow=all
allow=ulaw,alaw
direct_media=no
from_domain=sip.voice.getsmartalex.com
rewrite_contact=yes

[smartalex]
type=identify
endpoint=smartalex
match=sip.voice.getsmartalex.com
```

Replace `YOUR_USERNAME` and `YOUR_PASSWORD` with values from AiSky.

## Step 3: extensions.conf dialplan

Add a context for inbound calls from AiSky:

```
[from-smartalex]
exten => _X.,1,NoOp(Inbound from AiSky: ${EXTEN})
exten => _X.,n,Goto(from-internal,${EXTEN},1)
```

Reload:

```
asterisk -rx "pjsip reload"
asterisk -rx "dialplan reload"
```

## Step 4: verify registration

```
asterisk -rx "pjsip show registrations"
```

Expected: `smartalex` showing `Registered`.

## Step 5: add extensions in AiSky

Settings, then PBX. Populate the directory.

## Step 6: link agent and test

Agent Studio, then Telephony tab. Link the trunk. Call, verify, transfer.

## Alternative: Native Extension Mode

The setup above has Asterisk register OUT to AiSky. The inverse, AiSky registers as an extension on YOUR Asterisk box, is also supported and often simpler when you already have Asterisk as your source-of-truth. See [Native Extension Mode](/telephony/native-extension-mode). The pjsip endpoint you provision for AiSky looks like any other softphone, username, password, codecs, transport. AiSky registers, sits in your `pjsip show endpoints`, and you route DIDs to its extension number through your normal dialplan.

## Asterisk-specific notes

* **SIP tracing**: `asterisk -rvvvv` then `pjsip set logger on`.
* **TLS transport**: add a `type=transport` section with `protocol=tls` and point to your cert bundle. Use port 5061.
* **Custom headers**: use `PJSIP_HEADER()` in dialplan if you need to inspect P-Asserted-Identity or other SIP headers on transferred calls.
* **Multiple AiSky trunks**: each trunk needs its own unique section names (append `-2`, `-3`, etc.).

## Next steps

<CardGroup cols={2}>
  <Card title="FreePBX" href="/telephony/freepbx">
    GUI on top of Asterisk.
  </Card>

  <Card title="Troubleshooting" href="/telephony/troubleshooting">
    Error reference.
  </Card>
</CardGroup>